24 feb 2007 kl. 11.07 skrev Pavel Jezek:
Olle E Johansson wrote:
23 feb 2007 kl. 12.42 skrev Steve Davies:
Hi,
In older versions of asterisk I used to be able to use
"incominglimit=1" to effectively disable call waiting on a specific
SIP channel (Where broken phones do not allow this on the handset
itself)
In 1.2.x this became "call-limit=1", but this prevents the phone
from
opening a 2nd line in order to transfer a call using attended
transfer. The WiKi suggests using SetGroup() etc, but this does not
cater for the case where you are Dialling several different phones
simultaneously.
You can still set one call-limit for the user and another for the
peer. The peer call-limit would be used to prevent call waiting
and the user limit could be set to a reasonable level so the phone
can do transfers.
it can be also usefull to use 'limitonpeer' option in sip.conf
with this, it would not be needed to define separate type=user and
type=peer for each phone,
instead define one type=friend and apply limitonpeers=yes
;limitonpeers=no ; Apply all call limits ("limit=")
only to peers, never
; to users. This improves handling
of call limits
; and device states in certain
situations. The user part
; of a type=friend will still be
affected by the call
; limit, but Asterisk will only use
one object for
; counting the simultaneous calls.
Well, yes. That option does not exist in 1.2, it's someting I have
implemented
in svn trunk. And in this particular case, different call limits on
the user
and the peer seemed useful.
/O
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